The threshold of audibility of phase noise in ADC and DAC clocks is a fairly contentious issue in the HiFi and audiophile world. Some sources claim that jitter is clearly audible at low levels, and some claim that high levels of jitter are inaudible. The literature describes several tests, many with conflicting results.

One of the chief difficulties in testing the audibility of jitter is that it requires a complex hardware setup, which means that many listeners would be required to be present for an time consuming (and expensive) on site test. Over the last couple of months I have been thinking about organising a distributed listening test to look at the audibility of jitter in audio applications, based on algorithms for simulating the effects of jitter on signals. These algorithms are fairly well described in RF and telecomms engineering literature, and would be interesting for comparison purposes.

The kind of thing I have in mind is this:Use samples which are accepted to sound good -> simulate jitter -> perform listening tests -> perform more tests at different levels of jitter depending on results

The purpose of this thread is to get ideas of the Hydrogenaudio community about performing these tests. Some of the things I would appreciate input on are:

Does anyone else have any additional actual published papers on this topic of the audibility of jitter or listening tests?

What journals cover this topic, if any?

My focus is to understand:a) given a synthetic jitter profile, is it audible using DBT?b) given a real jitter environment, is it audible using DBT?c) can you DBT the difference between toslink and coax?

I have been looking into the basis of the audiophile belief that toslink is broken due to too much jitter and the implicit belief of the audibility of very small amounts of jitter. Given the intensity of the belief, perhaps where there is smoke, there is fire, even though this belief makes no sense to me based on my understanding of jitter and its impacts in this application space.

As far as I understand jitter impacts two things related to audio reproduction:a) at the level of the synchronous bit transport, it influences bit error ratesb) at the level of the DAC, it causes errors in the recostruction of the waveform. The outcome of this is essentially higher noise and distortion, i.e. you just get a bump in the noise floor related to the waveform being reconstructed - its correlated to what is being reconstructed, which is a wrinkle - and possibly spectral aliases being created.

Detailed studies of this understanding are also appreciated with actual waveforms & spectrum views. I also want to be clear I understand the techniques to dejitter a clock, including reclocking and most importantly, buffering. The issue is about impact, not about repair or avoidance.

I don't want any more audiophile "received wisdom" on the issue of jitter, I have received lots of that.

Papers without audibility studies set the threashold far lower. For example, Dunn's 1992 AES paper claims an audibility threshold of an astonishing 20ps at 20 KHz, based on his 1991 paper "Considerations for Interfacing Digital Audio Equipment to the Standards AES3, AES5, AES11, Proceedings of the 10th International AES Conference, 1991" (paper not yet found online). As another data point, "A Digital Discourse, Dr. Malcolm Hawksford; HiFi News & Record Review Feb,April, June, Aug, 1990" claims a peak jitter threshold of 400ps (also cited by Stereophile). I have not found the actual article yet, just citations and quotes.

Is Dunn's audibility curve an analytic derivation, or an audibility study? Dunn's curve of audibility is widely quoted. Anyone have a copy of this paper?

Others cited, but not yet found (I hesitate to pay the $20 AES paper fee) include "Eric Benjamin and Benjamin Gannon, "Theoretical and Audible Effects of Jitter on Digital Audio Quality", Preprint 4826 of the 105th AES Convention, San Francisco, September 1998" and "The Effects of Sampling Clock Jitter on Nyquist Sampling Analog-to-Digital Converters, and on Oversampling Delta-Sigma ADCs, 87th Convention of the Audio Engineering Society, October, 1989" (also cited by Stereophile).

There appears to be tremendous discussion of jitter measurement, but little understanding of what it actually means. A lot of this appears to me to be very old work and at best analytic, not audability based. None of it considers modern techniques to break the end to end synchronous clocking paradigm although some of them hint at what is now common practice in the telecommunications & Internet space.

Journals that publish in this area or references to further studies would be welcome.

(Zster, thanks for the reference to the Lavry overview paper. It is a useful overview document to help explain the impact of jitter in a well written way, and review some of the more modern methods to dejitter signals.)

Papers without audibility studies set the threashold far lower. For example, Dunn's 1992 AES paper claims an audibility threshold of an astonishing 20ps at 20 KHz, based on his 1991 paper "Considerations for Interfacing Digital Audio Equipment to the Standards AES3, AES5, AES11, Proceedings of the 10th International AES Conference, 1991" (paper not yet found online). As another data point, "A Digital Discourse, Dr. Malcolm Hawksford; HiFi News & Record Review Feb,April, June, Aug, 1990" claims a peak jitter threshold of 400ps (also cited by Stereophile). I have not found the actual article yet, just citations and quotes.

Is Dunn's audibility curve an analytic derivation, or an audibility study? Dunn's curve of audibility is widely quoted. Anyone have a copy of this paper?

Others cited, but not yet found (I hesitate to pay the $20 AES paper fee) include "Eric Benjamin and Benjamin Gannon, "Theoretical and Audible Effects of Jitter on Digital Audio Quality", Preprint 4826 of the 105th AES Convention, San Francisco, September 1998" and "The Effects of Sampling Clock Jitter on Nyquist Sampling Analog-to-Digital Converters, and on Oversampling Delta-Sigma ADCs, 87th Convention of the Audio Engineering Society, October, 1989" (also cited by Stereophile).

250ns and 400ps are not contradictory. One is saying "we've tested it subjectively - at around 250ns it starts to become audible". The other is saying "from first principles, if you keep it below 400ps, for the most sensitive possible signals, it will have a smaller impact on the signal than the limits of the system itself (i.e. the sample rate/bandwidth and bitdepth)".

These are two different approaches to audio engineering. One says "we can make it as bad as we want as long as no one can hear it". The other says "we will make it as good as we can to the point where this part can never be the limiting factor".

The real world has to sit between the two. You can't engineer something so that nothing is the limiting factor! You have to have some understanding of human ears to know when to stop improving everything (or be permanently depressed that nothing is good enough).

Conversely, you can't make everything "as bad as it can be before it causes an audible problem" because if you chain all these separate things together you can be fairly sure that you will have an audible problem at the end!

I like the idea of an experiment, but I don't see how a typical HA public test can work or be valid. We're all listening with unknown levels of jitter.

It would be like testing the audibility of -120dB of noise while we're all listening with soundcards which add noise at somewhere between -108dB and -60dB. Not hearing the -120dB of noise through these sound cards proves nothing. The same is true of jitter.